Method and system for securely authorizing VoIP interconnections between anonymous peers of VoIP networks

ABSTRACT

A peering authority or settlement clearinghouse can be used to control access, collect session accounting information, and provide financial settlement of interconnect or session fees among anonymous Internet Protocol (IP) peers or networks. The addition of peering policy criteria, such as price and quality of service, to peer to peer route discovery mechanisms enable a trusted intermediary, such as the settlement clearinghouse, to authorize acceptable interconnection or peering sessions between anonymous IP peers. Any financial settlement transactions which result from the peering sessions may be subsequently executed by the settlement clearinghouse.

PRIORITY CLAIM TO PROVISIONAL AND NON-PROVISIONAL APPLICATIONS

The present application claims priority to provisional patentapplication entitled, “Settlement Service for DUNDi Type VoIP Networks,”filed on Dec. 13, 2004 and assigned U.S. Application Ser. No.60/635,621; the entire contents of which are hereby incorporated byreference.

TECHNICAL FIELD

The invention relates to video, voice, data communications andapplication services. More particularly, the invention relates to asystem and method for securely authorizing VoIP interconnection accesscontrol between anonymous peers of VoIP networks.

BACKGROUND OF THE INVENTION

In the traditional telephone carrier operating model, calls betweenLocal Exchange Carriers (LECs), or Retail Service Providers (RSPs) aretransported by an Inter-Exchange Carrier (IXC). The RSP provides retailtelephone services to its end user subscribers on its network. When aRSP end user subscriber calls a telephone number which is not in theRSP's network, the RSP will switch that call to an IXC that willtransport the call to the RSP serving the called number to complete thetelephone call to the receiving party. The business model for this callscenario starts with the source RSP that switches the call to the IXC.The RSP pays the IXC a fee to transport the call to the destination RSP.The IXC transports the call to the destination RSP which completes thecall to the receiving party.

The IXC pays the destination RSP a fee to complete the telephone call.An important operating value added by the IXC is route discovery. TheIXC manages a central routing table that enables routing among amultitude of RSPs to any telephone number on the global Public SwitchedNetwork (PSTN). This action simplifies operations for the RSP operatorwhom needs to route to only one IXC to obtain termination to anytelephone number in the PSTN. In this document, the operating modeldescribed above is referred to as the IXC operating model.

This common telephony business model for the operating model describedabove is referred to as the Calling Party Pays model. The end user ofthe source RSP pays a retail service fee to the source RSP. The sourceRSP pays the IXC a fee to locate and transmit the call to thedestination RSP. The IXC pays the destination RSP a termination fee tocomplete the call. An important aspect of this business model is therole of the IXC as the central routing and billing intermediary amongmany RSPs. Source and destination RSPs do not have commercialinterconnect agreements with one another.

An important commercial value added by the IXC is the clearing of calls(routing and access control) between RSPs, accounting of interconnectedcalls and settlement of interconnect fees to ensure the destination RSPreceives a share of the revenue compensation as expected in the CallingParty Pays business model. Each RSP has a single bilateral interconnectagreement with the IXC which eliminates the costly need for commercialbilateral agreements with every other RSP.

Relative to the conventional IXCs, a new communications model hasevolved: The increasing use of Voice over IP (VoIP) communications hasmade possible a new operating model referred to as the Peer To Peeroperating model. The Peer To Peer operating model differs from the IXCoperating model because end to end routing and signaling for telephonecalls is achieved directly from the source RSP (peer) to the destinationRSP (peer) without the need for a central intermediary such as an IXC.Two examples of the Peer To Peer operating model are DUNDi and ENUM.DUNDi (Distributed Universal Number Discovery, www.dundi.com) enablessource networks (peers) to discover routes to destination networks(peers) without the need for a central routing directory or intermediarysignaling point.

ENUM is the Internet Engineering Task Force (www.itef.org) protocol (RFC2916) which defines how a source peer may resolve telephone numbers intoIP addresses in order to route and signal a VoIP call directly to thedestination network (peer). In other words, ENUM is a standard adoptedby the Internet Engineering Task Force (IETF) that uses the domain namesystem (DNS) to map telephone numbers to Web addresses or uniformresource locators (URL). The goal of the ENUM standard is to provide asingle number to replace the multiple numbers and addresses for anindividual's home phone, business phone, fax, cell phone, and e-mail.

However, while IP technology has enabled the Peer To Peer operatingmodel, there is no scalable mechanism to implement the Calling PartyPays business model with a Peer To Peer operating model. With the PeerTo Peer operating model, the Calling Party Pays business model can onlybe implemented if every RSP (peer) has a bilateral commercialinterconnect agreement with every other RSP (peer). Bilateral agreementsamong RSPs is not practical because the number of commercial peeringagreements for all RSPs increases by the square of the number of RSPs(peers) [n*(n−1)/2 where n=number of peers], making large scale peer topeer networks using the Calling Party Pays business model virtuallyimpossible.

Referring now to FIG. 1a , this figure illustrates a VoIP call within aRSP's network. Circle 100 in FIG. 1a represents the RSP network. The RSPnetwork could be a private IP network or a subset of public Internet.The call control point 110 controls calls between the calling andreceiving parties by providing calling party authentication, additionalservice features such as call forwarding, call signaling to thereceiving party and generating called detail records to account for thecall transaction. One of ordinary skill in the art who is familiar withVoIP technology will recognize that the Call Control Point could beeither an H.323 gatekeeper, H.323 IP to IP gateway, SIP proxy, SIP backto back user agent, softswitch, session border controller or any otherdevice which controls routing or signaling between source anddestination VoIP devices. Two end user subscribers of the RSP Networkare represented by a first telephone 120 with number 14045266060 andsecond telephone 130 with number 14045724600.

FIG. 1a represents a call scenario where the calling party 120 calls areceiving party 130. The call from the calling party 120 is initiatedwith a call setup message 400, such as a SIP Invite message to the CallControl Point 110. The Call Control Point determines if, and how, thecall should be routed to the receiving party 130. To complete the callto the receiving party 130, the Call Control Point 110, sends a message410 to the receiving party 130 to complete the call between the callingand called parties. When the RSP provides service to both the callingand called parties, the call can be completed within the RSP's network100 without the use of facilities provided by another VoIP serviceprovider. In FIG. 1a , inter-IP network peering does not occur.

Referring now to FIG. 1b , this Figure illustrates a VoIP call thatrequires inter-IP network peering. FIG. 1b includes the Source RSPNetwork 100, and with elements 110, 120 and 130 that are similar thosedescribed in FIG. 1a . Destination RSP Network 200 with Call ControlPoint 210 is introduced in FIG. 1b . Two end user subscribers of theDestination RSP Network 200 are represented by a third telephone 220with number 17033089726 and fourth telephone 230 with number17036054283. The calling party 120 places a call to telephone number17036054283. The call starts with a call setup message 400 from thecalling party 120 to the Call Control Point 110 of the Source RSPNetwork 100. The Source RSP Network 100 cannot complete the call withinits network, since the receiving party 230 is served by the DestinationRSP Network 200. Therefore, the source Call Control Point 110, sends amessage 420 to the Call Control Point 210 of the Destination RSP Network200. The destination Call Control Point 210 then sends a message to thereceiving party 230 to complete the call.

Completion of the call scenario in FIG. 1b requires peering between theSource RSP 100 and Destination RSP 200 networks. Peering between VoIPnetworks requires two functions. First, the source IP network 100 mustknow which destination VoIP network 200 can complete the call. Thisinformation is referred to as routing—the source network 100 must knowto which IP address the VoIP call should be routed. Routing informationcan be pre-programmed into the Call Control Point of the source networkbased on a pre-arranged, bilateral peering agreement between source anddestination networks, or discovered in real time using mechanisms suchas DUNDi or ENUM referred to previously.

The second function required for peering is access permission. Thesource network 100 must be permitted to access the destination network200 to complete the call. Access permission between two IP networks iscommonly controlled by the use of an access list at the destinationnetwork. The destination network 200 will only accept IP communicationsfrom IP addresses in its access control list. Other access controltechniques are based on the inclusion of a password or digital signaturein the call setup message 420 between the source and destinationnetworks. If the destination network 200 can validate that the passwordor digital signature can only be from a trusted source, the call orpeering transaction can be accepted without the source IP address beingincluded in an access control list.

There are several limitations with this conventional technology used forVoIP interconnection or peering. First, the technique of bilateralpeering agreements is difficult to implement when a large number ofbilateral peering agreements must be maintained. Real time routediscovery techniques such as ENUM or DUNDi provide scalable solutionsfor inter-peer routing but do not provide scalable mechanisms forinter-peer access control or accounting. Accordingly, there is a need inthe art for a scalable. technique for inter-peer access control andaccounting that is independent of the route discovery mechanism. Afurther need exists for a reliable scalable mechanism for implementingthe Calling Party Pays business model with a Peer to Peer operatingmodel:

A need exists in the art to solve this scalability problem for theCalling Party Pays business model in a Peer To Peer operating model. Aneed also exists in the art for eliminating or substantially reducingthe number of bilateral agreements among RSPs.

SUMMARY OF THE INVENTION

According to one exemplary aspect of the technology, RSPs using VoIP mayestablish a single bilateral commercial agreement with a trusted thirdparty or clearinghouse that can authorize interconnection, on a call bycall basis, among source and destination RSPs. These source anddestination RSPs will typically not have bilateral commercialinterconnect agreements. The invention can comprise a trusted settlementclearinghouse that ensures interconnect data such as calling number,caller ID, interconnect rates and other critical data are valid and thenexecutes any resulting financial transactions between the source anddestination RSPs.

Exemplary aspects of the invention will refer to interconnections amongRSPs providing VoIP telephony services. However, one of ordinary skillin the art will recognize that this invention can be used for peeringaccess control and accounting between IP networks on a session bysession basis for many applications in addition to VoIP, such as videosessions, data transfers with a guaranteed quality of service, bandwidthreservation, conferencing of three or more telephony or video sessions,content brokering, short message services, gaming and instant messaging.

A settlement clearinghouse, according to one exemplary aspect of theinvention, can be referred to more generally as a Peering Authority, andmay be used for peering access control and accounting of other IPapplications in addition to VoIP. A settlement clearinghouse or peeringauthority can comprise a common trusted third party for all peers. Thesettlement clearinghouse can exchange digital certificates with eachpeer and use asymmetric key cryptography to establish and manage atrusted, bilateral relationship with each peer. These trusted bilateralrelationships between each peer and the settlement clearinghouse canenable the settlement clearinghouse to securely authorize VoIPinterconnection access control between anonymous peers on a call by callbasis. In addition, the settlement clearinghouse can also securelycollect accounting information for each call interconnected between VoIPnetworks. This accounting information may then be used for the trackingor billing of interconnected VoIP calls and execution of inter-networkfinancial settlements.

According to another exemplary aspect of the invention, a source IPnetwork may specify routing and all terms of an individual peeringsession with the destination network of its choice. According to otherexemplary aspects of the technology, the inventive system and methoddescribes how a trusted third party clearinghouse, or peering authority,can provide a centralized and scalable for solution for authorizing andaccounting for inter-network peering sessions among known and anonymouspeers. Exemplary aspects of the inventive system also include thediscrete elements that form each of the individual peering authorizationrequest messages, peering authorization response messages, and thepeering authorization tokens. The discrete elements of the messages andauthorization tokens are described in further detail below.

According to a further exemplary aspect, the inventive system comprisesa technique that can decouple IP peering access control and accountingfrom routing. The inventive system illustrates how a source IP peer cansubmit a peering request to a trusted Peering Authority for accessauthorization to a known destination peer. Unlike conventional routingrequests that are used by source networks to find the route to adestination, the peering request can comprise routing and all commercialterms (price, type of service, quality of service) for the proposedpeering session. The role of the Peering Authority is to authorize andaccount for the peering transaction between the source and destinationpeers which have no trusted or commercial relationship.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1a is a functional block diagram that illustrates a conventionalVoIP call within a RSP's network.

FIG. 1b is a functional block diagram illustrating a conventional VoIPcall that requires inter-IP network peering.

FIG. 2a is a functional block diagram illustrating an exemplary callscenario according to one exemplary embodiment of the invention.

FIG. 2b is a functional block diagram illustrating how call detailrecords are collected by a settlement clearinghouse for interconnectaccounting and settlement billing according to one exemplary embodimentof the invention.

FIGS. 3a-3d are logic flow diagrams illustrating a process of how aclearinghouse or peering authority authorizes and tracks accountinginformation for a VoIP call interconnected between two IP networksaccording to one exemplary embodiment of the invention.

FIG. 4a is a functional block diagram illustrating an inter-IP networkpeering scenario that includes a peering authorization request accordingto an exemplary embodiment of the invention.

FIG. 4b is a functional block diagram illustrating an exemplary peeringauthorization response message according to one exemplary embodiment ofthe invention.

FIG. 4c is a functional block diagram illustrating an exemplary callsetup message, with peering authorization token, between peers accordingto one exemplary embodiment of the invention.

FIG. 4d is a functional block diagram illustrating exemplary peeringaccounting messages according to one exemplary embodiment of theinvention.

DETAILED DESCRIPTION OF EXEMPLARY EMBODIMENTS

A settlement clearinghouse or peering authority can comprise a commontrusted third party for all peers of VoIP networks. The settlementclearinghouse can exchange digital certificates with each peer and useasymmetric key cryptography to establish and manage a trusted, bilateralrelationship with each peer. These trusted bilateral relationshipsbetween each peer and the settlement clearinghouse can enable thesettlement clearinghouse to securely authorize VoIP interconnectionaccess control between anonymous peers of VoIP networks on a call bycall basis. In addition, the settlement clearinghouse can also securelycollect accounting information for each call interconnected between VoIPnetworks. This accounting information may then be used for the trackingor billing of interconnected VoIP calls and execution of inter-networkfinancial settlements.

Referring now to the drawings, in which like numerals represent likeelements throughout the several Figures, aspects of the invention andthe illustrative operating environment will be described.

An exemplary call scenario can begin with a calling party 120 who callsa telephone number 17036054283 as illustrated in FIG. 2a . The receivingparty with this telephone number is denoted with reference numeral 230.A call setup message 400 is sent from the calling party 120 to the CallControl Point 110 of the Source RSP network. In this call scenario, theCall Control Point 110 of the source network 100 knows the IP address ofthe destination network 200 that can complete the call and theinterconnect price the destination network 200 will charge to completethe call.

This interconnect routing and rate information may have beenpre-configured based on a bilateral agreement between the source 100 anddestination networks 200 or may have been discovered in real time usingsome other mechanism. However, before sending call setup message 440,the source Call Control Point 110 sends an interconnect authorizationrequest message 310 to the Settlement Clearinghouse 300. One of ordinaryskill in the art of IP communications recognizes that messages to andfrom the Settlement Clearinghouse 300 may be encrypted to ensure themessage contents are secure.

The Clearinghouse 300 may operate in a networked environment usinglogical connections to one or more other remote computers. TheClearinghouse 300 and Call Control Points 110, 210 can comprisecomputers such as a personal computer, a server, a router, a network PC,a peer device, or a common network node. The logical connectionsdepicted in FIG. 2a can include additional local area networks (LANs)and a wide are networks (WANs) not shown. Such networking environmentsare commonplace in offices, large industrial facilities, enterprise widecomputer networks, intranets, and the Internet. While conventionaltelephones 120, 130, 220, and 230 are illustrated in each of theFigures, one of ordinary skill in the art recognizes that thesetelephones can comprise electronic devices that support VoIP. Forexample, the telephones 120, 130, 220, and 230 can comprise a generalpurpose computer connected to respective networks 100, 200.

The Clearinghouse 300 and Call Control Points 110, 210 illustrated inFIG. 1 may be coupled to a LAN through a network interface or adaptor.When used in a WAN network environment, the computers may typicallyinclude a modem or other means for establishing direct communicationlines over the WAN. In a networked environment, program modules may bestored in remote memory storage devices. It will be appreciated that thenetwork connections shown are exemplary and other means of establishinga communications link between computers other than depicted may be used.

Moreover, those skilled in the art will appreciate that the presentinvention may be implemented in other computer system configurations,including other hand-held devices besides hand-held computers,multiprocessor systems, microprocessor based or programmable consumerelectronics, networked personal computers, minicomputers, mainframecomputers, and the like.

The invention may be practiced in a distributed computing environment asillustrated in FIG. 2a , where tasks may be performed by remoteprocessing devices that are linked through a communications network suchas the distributed computer networks 100, 200. In a distributedcomputing environment, program modules may be located in both local andremote storage devices.

The illustrated telephones 120, 130, 220, and 230 can comprise anygeneral purpose computer capable of running software applications. Thetelephones 120, 130, 220, and 230 can be portable for mobileapplications and they may be coupled to the respective networks 100, 200though wired or wireless links. Typical wireless links include a radiofrequency type in which the telephones 120, 130, 220, and 230 cancommunicate to the respective networks 100, 200 using radio frequency(RF) electromagnetic waves. Other wireless links that are not beyond thescope of the invention can include, but are not limited to, magnetic,optical, acoustic, and other similar wireless types of links.

Referring again to FIG. 2a , the interconnect authorization requestmessage 310 can also be referred to more generally as a peeringauthorization request and may be used for other IF applications inaddition to VoIP. The authorization request message 310 may include nameand identification information about the source and destinationnetworks, the calling and called parties and the interconnect rate forthe call between the two networks.

To implement the Calling Party Pays business model, a positiveinterconnect rate indicates that the Source RSP Network 100 will pay theDestination RSP Network 200 to complete the call to the receiving party.The Settlement Clearinghouse 300, acting as the trusted third partybetween the Source RSP Network 100 and the Destination RSP Network 200will approve or reject the interconnect authorization request message310, based on the interconnect policies enforced by the SettlementClearinghouse 300.

The Settlement Clearinghouse 300 responds to an interconnectauthorization request message 310 by sending an authorization responsemessage 315 back to the source Call Control Point 110 indicating thatthe authorization request was approved or rejected. The interconnectauthorization response message 315 may also be referred to moregenerally as the peering authorization response and may be used forother IF applications in addition to VoIP. If the interconnectauthorization request message 310 is approved by the SettlementClearinghouse 300, the interconnect authorization response message 315will comprise an interconnect authorization token that is returned tothe Source RSP Network 100. The interconnect authorization token willalso be referred to more generally as the peering authorization tokenand may be used for other applications in addition to VoIP.

The Settlement Clearinghouse 300 will typically sign the interconnectauthorization token with its digital signature to ensure non-repudiationof the authorization token and to guarantee that the SettlementClearinghouse 300 is party to the interconnection or peering transactionbetween the Source RSP Network 100 and the Destination RSP Network 200.

Another valuable service which may be provided by the SettlementClearinghouse 300 is authentication and verification of the name andidentification of either, or both, of the calling party 120 and theSource RSP Network 100. This function is especially useful for theDestination RSP Network 200 and receiving party 230 when a call isreceived from an unknown source network or anonymous peer. If the nameand identification of either, or both, of the calling party 120 andsource network 100 have been verified by the Settlement Clearinghouse300 and are included in the signed authorization token. conveyed in theinterconnect call setup message 440, the destination network 200 andreceiving party 230 may have some assurance that the name andidentification information is legitimate.

When the source Call Control Point 110 receives interconnectauthorization approval in the response 315 from the SettlementClearinghouse 300, it can extract the interconnect authorization tokenfrom the response 315 and insert the authorization token in the callsetup message 440 to the Call Control Point 210 of the Destination RSPNetwork 200. The destination Call Control Point 210 reviews theinterconnect authorization token contained in the call setup message 440to determine if it is valid.

Determining if the interconnect authorization token valid can beaccomplished by the Call Control Point 210 validating the digitalsignature of the signed authorization token. If the interconnectauthorization token has been signed by a trusted third party, such as bythe Settlement Clearinghouse 300 who may have a bilateral commercialinterconnect agreement with Call Control Point 210, then the token isvalid and the call will be accepted, even if the call originates from anunknown IP address or anonymous peer. The token is deemed valid becauseof the relationship between the Settlement Clearinghouse 300 and CallControl Point 210.

To implement the Calling Party Pays business model, the signedauthorization token in the call setup message 440 will include theinterconnect rate required by the Destination RSP Network 200. Thesigned token with the interconnect rate, provides the Destination RSPNetwork 200 with a document that cannot be repudiated or rejected by theSettlement Clearinghouse 300. The interconnect authorization tokencontained in the setup message 440 is evidence that the SettlementClearinghouse approved the interconnection between the source anddestination networks at the specified rate.

FIG. 2b illustrates how call detail records are collected by theSettlement Clearinghouse 300 for interconnect settlement billing. Whenthe call between the calling party 120 and receiving party 230 ends, thesource Call Control Point 110 sends a call detail record 320 to theSettlement Clearinghouse 300 and the destination Call Control Point 210also sends a call detail record 330 to the Settlement Clearinghouse 300.The call detail records 320 and 330 may include the interconnect rateapproved by the Settlement Clearinghouse 300 in the interconnectauthorization token.

One of ordinary skill in the art of IP communications will recognizethat the technique described above for inter-IP network access controland accounting for VoIP applications can also be applied generally formany other IP applications that require the use or facilities ofmultiple networks. For example, exchanging video programs over the IPnetwork using the described inter-IP network access control is notbeyond the scope of the invention.

Exemplary Process for Securely Authorizing VoIP Interconnections BetweenAnonymous Peers

The processes and operations of the inventive system described belowwith respect to all of the logic flow diagrams may include themanipulation of signals by a processor and the maintenance of thesesignals within data structures resident in one or more memory storagedevices. For the purposes of this discussion, a process can be generallyconceived to be a sequence of computer-executed steps leading to adesired result.

These steps usually require physical manipulations of physicalquantities. Usually, though not necessarily, these quantities take theform of electrical, magnetic, or optical signals capable of beingstored, transferred, combined, compared, or otherwise manipulated. It isconvention for those skilled in the art to refer to representations ofthese signals as bits, bytes, words, information, elements, symbols,characters, numbers, points, data, entries, objects, images, files, orthe like. It should be kept in mind, however, that these and similarterms are associated with appropriate physical quantities for computeroperations, and that these terms are merely conventional labels appliedto physical quantities that exist within and during operation of thecomputer.

It should also be understood that manipulations within the computer areoften referred to in terms such as listing, creating, adding,calculating, comparing, moving, receiving, determining, configuring,identifying, populating, loading, performing, executing, storing etc.that are often associated with manual operations performed by a humanoperator. The operations described herein can be machine operationsperformed in conjunction with various input provided by a human operatoror user that interacts with the computer.

In addition, it should be understood that the programs, processes,methods, etc. described herein are not related or limited to anyparticular computer or apparatus. Rather, various types of generalpurpose machines may be used with the following process in accordancewith the teachings described herein.

The present invention may comprise a computer program or hardware or acombination thereof which embodies the functions described herein andillustrated in the appended flow charts. However, it should be apparentthat there could be many different ways of implementing the invention incomputer programming or hardware design, and the invention should not beconstrued as limited to any one set of computer program instructions.

Further, a skilled programmer would be able to write such a computerprogram or identify the appropriate hardware circuits to implement thedisclosed invention without difficulty based on the flow charts andassociated description in the application text, for example. Therefore,disclosure of a particular set of program code instructions or detailedhardware devices is not considered necessary for an adequateunderstanding of how to make and use the invention. The inventivefunctionality of the claimed computer implemented processes will beexplained in more detail in the following description in conjunctionwith the remaining Figures illustrating other process flows.

Further, certain steps in the processes or process flow described in allof the logic flow diagrams below must naturally precede others for thepresent invention to function as described. However, the presentinvention is not limited to the order of the steps described if suchorder or sequence does not alter the functionality of the presentinvention. That is, it is recognized that some steps may be performedbefore, after, or in parallel other steps without departing from thescope and spirit of the present invention.

Referring now to FIG. 3, the logical flow charts of FIGS. 3a, 3b, 3c and3d illustrate the process of how a clearinghouse, or peering authority,is used to authorize and track accounting information for a VoIPcall/communication exchanged between two IP networks. The inter-networkcall scenario starts in FIG. 3a in Step 002 when the calling party, anend user on the source network 100, initiates a call to a receivingparty on an external network 200. In Step 002, a call setup message 400(as illustrated in FIG. 2a ) is created and is sent to the call controlpoint 110 of the source network 100. Call control point 110 in Step 004then determines that the call cannot be completed on the local networkand must be routed to the external network serving the receiving party230. Interconnecting with an external network will require permission toaccess the external network. This is the stage in the process where apeering authority or clearinghouse can play a role.

Step 006 can comprise two sub-steps: In the first sub-step, the sourcenetwork 100 usually must determine which external network(s) serve(s)the receiving party 230 (sometimes through using route discovery) anddetermine the terms of interconnecting with the external network(peering criteria). There are many established ways the source network100 can determine the how the call can be routed to the destinationnetwork 200 serving the receiving party 230.

Routes to the external called number of the receiving party 230 could bepre-programmed in the routing table of the call control point 110 of thesource network 100 based on negotiated interconnect agreements withdestination networks 230. Alternatively, the routes can be discovered inreal time using protocols such as ENUM or DUNDi. Once the call controlpoint 110 of the source network 100 has determined the possible routesto the receiving party 230, the second sub-step of Step 006 is for thecall control point 110 of the source network 100 to determine additionalpeering criteria such as bandwidth, network quality of service, and theprice the calling party 120 must pay the destination network 200 tocomplete the call.

The peering criteria information can be based on interconnect agreementsnegotiated with destination networks 200 or advertising. In the future,the peering criteria may be obtained from IP protocols that advertisepeering prices and service levels over the network. When the route andpeering criteria are determined, this information is then sent in Step008 as a peering authorization request 310 to the Clearinghouse 300 asillustrated in FIG. 2 a.

After the Clearinghouse 300 receives the peering authorization requestfroth the call control point 110 of the source network 100, in Step 010the Clearinghouse 300 authenticates the source of the information andthe calling party. Specifically, in step 012, the Clearinghouse 300authenticates the source device (call control point 110) which sent thepeering request. Step 012 can be completed in various ways such aschecking the IP address of the source device or more securely usingSecure Sockets Layer (SSL) client authentication. If the source devicecannot be authenticated, the peering request may be denied in Step 014.Identifying the source device will reveal the operator of the sourcenetwork that has established a trusted relationship with theClearinghouse.

Next, in step 016, the Clearinghouse 300 checks the status of the sourcenetwork operator to determine if it may originate calls or be grantedaccess to another network. If access to another network is not allowed,the peering request may be denied in Step 018. As part of this process,the Clearinghouse 300 may determine details about the source networkidentification, such as the organization name and address. One importantelement in the source network identification is information whichindicates how the source network identification was verified.

This information about the source network 100 that is verified by theClearinghouse 300 can be a value added service for the receiving party200. In Step 020, the Clearinghouse 300 may check the status of thecalling party to determine if it may access external networks. TheClearinghouse 300 may also take actions to identify the calling party.Step 020 can be similar to a Caller Name (CNAM) look-up in thetraditional Public Switched Telephone Network (PSTN) or some othermechanism which more securely identifies and verifies the calling partyidentification. If the calling party is not allowed to access networksexternal to the source network, the peering request may be denied inStep 022.

Referring now to FIG. 3b , after the peering request has been fullyauthenticated, the next step for the Clearinghouse 300 is to examine thepeering criteria in the peering request for correctness in Step 024.Peering requests that are illogical, impossible or which cannot bebilled for may be denied. Specifically, in Step 026, the Clearinghouse300 can determine if the type of service requested is possible. Forexample, if the peering request specifies a video session and thedestination peer does not support video, then the peering request willbe denied in Step 028.

Next, in Step 030, the Clearinghouse 300 can check the pricing terms ofthe peering request. The pricing terms of the peering request can becompared to pricing tables stored in the Clearinghouse 300. If thepricing terms of the request do not match any entries of the table(s),or if the pricing terms are incomplete or ambiguous, the peering requestmay be denied in Step 032. For example, if the currency specified isJapanese Yen (JPY) and the clearinghouse only performs settlement in USDollars (USD) then the peering request would be denied in Step 032.

In Step 034, the Clearinghouse 300 checks if the quality of service(QoS) terms of the peering request. The Clearinghouse 300 can check theQoS terms of the peering request against stored values in tables thatthe Clearinghouse may have for the destination networks 200. If theservice level is not supported, then the peering request will be deniedby the Clearinghouse in Step 036. For example, if a peering requestspecifies 64 kb/sec bandwidth for a VoIP call and the Clearinghouse 300recognizes that 64 kb/sec bandwidth cannot be provided by thedestination network 200, then the peering request will be denied in Step036.

In Step 038, the Clearinghouse 300 compares historical quality ofservice of the destination device or network 200 to the quality ofservice requested in the peering request. If the historical quality ofservice is less than the requested quality of service, the peeringrequest may be denied in Step 040. For example, if the Answer SeizureRatio specified in the peering request is 50%, but Clearinghousehistorical records indicate that the destination device or network 200has a historical Answer Seizure Ratio of 40% then the peering requestwould be denied in Step 040.

If all authentication and peering criteria checks are successful, theClearinghouse 300 creates a peering authorization token for eachdestination network 200 in Step 042. The token is usually digitallysigned using a private key of the Clearinghouse 300 to ensure dataintegrity and non-repudiation of the token. The tokens are then returnedto the call control point 110 of source network 100 in a peeringauthorization response 315 in Step 044.

Referring now to FIG. 3c , when the call control point 110 of sourcenetwork 100 receives the peering authorization response 315 from theClearinghouse 300, in Step 046, the call control point 110 can selectthe first destination network 200 to complete the call. In step 048, thecall control point 110 forms the call setup message 440 so that itcomprises the peering authorization token.

When the destination network 200 receives the call setup message, inStep 050 the call control point 210 will validate the peering tokencontained in the setup message before accepting the call. A commonpractice for securely validating tokens is validating the digitalsignature of the token using the public key of the clearinghouse 300. InStep 052, the token can be validated using the public key. If thedigital signature is valid; then the destination network 200 can becertain that the token was signed using the Clearinghouse private key.If the token is not valid, the call control point 210 of the destinationnetwork 200 will block the call in Step 054. The process then continuesin Step 068 in which the call control point 110 of the source networkselects the next available destination network 200. If the inquiry todecision Step 052 is positive, then the process proceeds to decisionStep 056.

If the token is valid, the call control point 210 of the destinationnetwork 200 may choose to check the peering criteria present in thetoken. The peering criteria can comprise service type, pricing terms,quality of service, just to name a few. Other peering criteria is notbeyond the scope of the invention. In decision Step 056, the callcontrol point 210 can determine if the service type present in the tokenis acceptable for its network configuration. If the inquiry to decisionStep 056 is negative, then the call is blocked in Step 058. The processthen continues in Step 068 in which the call control point 110 of thesource network selects the next available destination network 200. Ifthe inquiry to decision Step 056 is positive, then the process proceedsto decision Step 060.

In decision Step 060, the call control point 210 can determine if thepricing terms found in the token are acceptable for its network terms.If the inquiry to decision Step 060 is negative, then the call isblocked in Step 062. The process then continues in Step 068 in which thecall control point 110 of the source network selects the next availabledestination network 200. If the inquiry to decision Step 060 ispositive, then the process proceeds to decision Step 064.

In decision Step 064, the call control point 210 can determine if thequality of service terms found in the token are acceptable for itsquality of service parameters. If the inquiry to decision Step 064 isnegative, then the call is blocked in Step 066. The process thencontinues in Step 068 in which the call control point 110 of the sourcenetwork selects the next available destination network 200.

Referring now to FIG. 3d , if the peering token validation processconducted by the call control point 210 of the destination network 200passes, the destination network completes the call to the receivingparty 230 (as illustrated in FIG. 2a ) in Step 070. When theconversation ends and the calling and receiving parties 120, 230 hang-upin Step 072, both the source and destination networks 100, 200 send callrecords 320, 330 to the Clearinghouse 300 in Step 074. The Clearinghouse300 uses the call record information to execute settlement procedures inStep 076. For example, the Clearinghouse 300 may perform variousservices on behalf of the peers (networks 100, 200) such as analyzingand reporting inter-peer traffic flows, billing for peering sessions, orexecution of any cash settlement among peers related to peering sessions076.

Peering Authorization Request 310

The Peering Authorization Request 310 from the source IP network 100 tothe Peering Authority (Clearinghouse 300) may include the followinginformation listed in Table 1 below.

TABLE 1 Peering Authorization Request Information Information ElementDescription Date and time Date/Time of the peering authorizationrequest. stamp Call Identifier Unique identifier for the call GroupIdSame as ConferenceID in H.323. Calls with unique CallIds can share acommon GroupID. i.e a conference call. Calling Party Unique identifierfor the calling party, i.e.: ITU E.164 telephone number, sip uri, teluri, IP address and port, name resolved by DNS to an IP address. CallingParty A set of information (Calling Party Name, Calling IdentificationParty Authority and Calling Party Verification) which identifies thecalling party. Calling Party Name or text description of the callingparty. For Name calling parties from the PSTN, this value wouldtypically be name from a Line Interface Database (LIDB) or Calling Name(CNAM) database that corresponds to the calling party's telephonenumber. Calling Party Organization of the calling party. OrganizationCalling Party First name of calling party. First Name Calling Party Lastname of calling party. Last Name Calling Party Street address of callingparty. Street Name Calling Party Street number of calling party. StreetNumber Calling Party Additional address information, such as apartmentAddress2 or suite of calling party. Calling Party Postal code of callingparty. Postal Code Calling Party City of calling party. City CallingParty State of calling party. State Calling Party Country of callingparty. Country Calling Party Unique number identifying the callingparty. Id Number Calling Party Website of calling party. Website CallingParty Uniform Resource Identifier of calling party. URI Calling PartyName or text description of the authority that Authority authenticatedand verified the calling party identification for the peeringauthorization request. Calling Party Integer value which indicates thelevel of Verification verification of the calling party identification.00 - No verification 01 - Based on the calling party's telephone number02 - Based on source device IP address 03 - Combination of 01 and 0204 - Based on password of calling party 05 - Combination of 01 and 0407 - Combination of 01, 02 and 04 08 - Based on the SSL/TLS clientauthentication of the calling party. If the Verification technique isunknown, then the Verification value should be empty. Source Network IPaddress and port (optional) or name resolved by Domain Name Server (DNS)to an IP address. Source Network A set of information (Source NetworkName, Source Identification Network Authority and Source NetworkVerification) which identifies the operator of the source network.Source Network Name or text description of the source network Nameoperator. Source Network Organization of the Source Network.Organization Source Network Street address of Source Network. StreetName Source Network Street number of Source Network. Street NumberSource Network Additional address information, such as apartmentAddress2 or suite of Source Network. Source Network Postal code ofSource Network. Postal Code Source Network City of Source Network. CitySource Network State of Source Network. State Source Network Country ofSource Network. Country Source Network Unique number identifying theSource Network. Id Number Source Network Website of Source Network.Website Source Network Uniform Resource Identifier of Source Network.URI Source Network Name or text description of the authority thatAuthority authenticated and verified the source network identificationfor the peering authorization request. Source Network Integer valuewhich indicates the level of Verification verification of the sourcenetwork identification. 00 - No verification 01 - Based on the callingparty's telephone number 02 - Based on source device IP address 03 -Combination of 01 and 02 04 - Based on password of calling party 05 -Combination of 01 and 04 07 - Combination of 01, 02 and 04 08 - Based onthe SSL/TLS client authentication of the calling party. If theVerification technique is unknown, then the Verification value should beempty. Source device IP address and port (optional) or name resolved byDNS. Source trunk String value with trunk group. This value may or groupmay not include circuit ID. Receiving party Unique identifier for thereceiving party, i.e.: ITU E.164 telephone number, sip uri, tel uri, IPaddress and port, name resolved by Domain Name Server (DNS) to an IPaddress. Application Application requested by the source network andserved by the destination network. This includes any applicationprovided as a service such as a gaming or video streaming from aspecific web camera. File File requested by the source network andserved by the destination network. This includes data files, ring tones,audio files or video files. Destination device IP address and port(optional) or name resolved by DNS. Destination trunk String value withtrunk group. This value may or group may not include circuit ID.Currency - Pricing Currency of billing rate, i.e USD. Indication Setup -Pricing Amount of currency - Fixed billing rate per call or Indicationtransaction. Amount - Pricing Amount of currency - Billing rate perincrement. Indication Increment - Pricing Number of units in eachbilling increment. Indication Unit - Pricing Seconds, packets, bytes,pages, calls. Indication Type of service voice, video, bandwidthreservation, conference. requested SubscriberInfo Data string whichidentifies calling party or subscriber, i.e. username and PIN.CustomerId Customer ID, identifies the Peering Authority customer oroperator who controls the source network. DeviceId Identifies the sourcedevice. Data Rate Data rate requested for VoIP call or IP session.Number of Channels Number of channels requested for IP session.Bandwidth Amount of bandwidth reserved for IP session. CodecCompression/decompression algorithm requested. Quality of Service Levelof service quality requested. Quality of Service Class of servicequality requested. Class Answer Seizure Minimum acceptable ASR. If thehistorical ASR for Ratio (ASR) calls to a destination device is lessthan the ASR in the peering request, the call should not be authorized.Mean Hold Minimum acceptable MHT. If the historical MHT Time (MHT) forcalls to a destination device is less than the MHT in the peeringrequest, the call should not be authorized. Post Dial Minimum acceptablePDD. If the historical PDD of Delay (PDD) calls to a destination deviceis less than the PDD in the peering request, the call should not beauthorized. Delay Minimum acceptable average one-way packet delay fortransmissions sent or received by a destination device. If the PeeringAuthority has historical delay statistics for destination device thatare greater than the delay value in the peering request, the call shouldnot be authorized. Jitter Minimum acceptable variance of packet delay(transit time from source to destination) for transmissions sent orreceived by a destination device. If the Peering Authority hashistorical jitter statistics for the destination device that are greaterthan the jitter value in the peering request, the call should not beauthorized. PackLoss Number of packets lost/total packets fortransmissions sent or received by a destination device. If the PeeringAuthority has historical PackLoss data for the destination device thatare greater than the PackLoss value in the peering request, the callshould not be authorized.Peering Authorization Request 310—XML Mapping

Table 2 below maps peering authorization request message informationelements to eXtensible Markup Language (XML) tags.

TABLE 2 Peering Authorization Request - XML TAGS Information Element XMLtag Date and time <Timestamp> stamp Call Identifier <CallIdencoding=″base64″> CallId may or may not be encoded. Calling party<SourceInfo type=″e164″> <SourceInfo type=″sip″> <SourceInfotype=″h323″> <SourceInfo type=″url″> <SourceInfo type=″email″><SourceInfo type=″transport″> <SourceInfo type=″international″><SourceInfo type=″national″> <SourceInfo type=″network″> <SourceInfotype=″subscriber″> <SourceInfo type=″abreviated″> <SourceInfotype=″e164prefix″> <SourceInfo type=″tel″> <SourceInfo type=″enum″>“transport” value must be an IP address or name resolved by DNS.type=″tel″ is a phone number in tel uri format. type=″enum″ is a phonenumber in ENUM format. Calling Party <CallingPartyId> XML tag indicatingthat Identification sub-tags <Name>, <Organization>, <FirstName>,<LastName>, <StreetName>, <StreetNumber>, <Address2>, <PostalCode>,<City>, <State>, <Country>, <IdNumber>, <Website>, <uri>, <Authority>and <Verification> correspond to the Calling party. Name <Name>Organization <Organization> First Name <FirstName> Last Name <LastName>Street Name <StreetName> Street Number <StreetNumber> Address2<Address2> Postal Code <PostalCode> City <City> State <State> Country<Country> Id Number <IdNumber> Website <Website> Uniform <uri> ResourceInd. Authority <Authority> Verification <Verification> Source Network<SourceAlternate type=″sip″> <SourceAlternate type=″url″><SourceAlternate type=″transport″> <SourceAlternatetype=″international″> <SourceAlternate type=″national″> <SourceAlternatetype=″network″> <SourceAlternate type=″abreviated″ > <SourceAlternatetype=″e164prefix″> “transport” value must be an IP address or nameresolved by DNS Source Network <SourceNetworkId> XML tag indicating thatIdentification sub-tags <Name>, <Organization>, StreetName>,<StreetNumber>, <Address2>, <PostalCode>, <City>, <State>, <Country>,<IdNumber>, <Website>, <uri>, <Authority> and <Verification> correspondto the source network or the network operator of the Calling party.Source device <DeviceInfo type=″e164″> <DeviceInfo type=″sip″><DeviceInfo type=″h323″> <DeviceInfo type=″url″> <DeviceInfotype=″email″> <DeviceInfo type=″transport″> <DeviceInfotype=″international″> <DeviceInfo type=″national″> <DeviceInfotype=″network″> <DeviceInfo type=″abreviated″> <DeviceInfotype=″e164prefix″> <DeviceInfo type=″tel″> <DeviceInfo type=″enum″>“transport” value must be an IP address or name resolved by DNS. “tel”value is a tel uri. Source trunk <SourceAlternate type=″network″> groupString value with trunk group. This value may or may not include circuitID. Receiving party <DestinationInfo type=″e164″> <DestinationInfotype=″sip″> <DestinationInfo type=″h323″> <DestinationInfo type=″url″><DestinationInfo type=″email″> <DestinationInfo type=″transport″><DestinationInfo type=″international″> <DestinationInfo type=″national″><DestinationInfo type=″network″> <DestinationInfo type=″subscriber″><DestinationInfo type=″abreviated″> <DestinationInfo type=″e164prefix″><DestinationInfo type=″tel″> <DestinationInfo type=″enum″> “transport”value must be an IP address or name resolved by DNS. “tel” value is atel uri. Application <Application> File <File> Destination<DestinationAlternate type=″e164″> network <DestinationAlternatetype=″sip″> <DestinationAlternate type=″h323″> <DestinationAlternatetype=″url″> <DestinationAlternate type=″email″> <DestinationAlternatetype=″transport″> <DestinationAlternate type=″international″><DestinationAlternate type=″national″> <DestinationAlternatetype=″subscriber″> <DestinationAlternate type=″abreviated″><DestinationAlternate type=″e164prefix″> <DestinationAlternatetype=″tel″> <DestinationAlternate type=″enum″> “transport” value must bean IP address or name resolved by DNS. “tel” value is a tel uri.Destination trunk <DestinationAlternate type=″network″> group Stringvalue with trunk group. This value may or may not include circuit ID.Amount - Usage <Amount> (within <UsageDetail> tags) Detail Increment -Usage <Increment> (within <UsageDetail> tags) Detail Unit - Usage <Unit>(within <UsageDetail> tags) Detail Currency - Pricing <Currency> (within<PricingIndication> tags) Indication Setup - Pricing <Setup> (within<PricingIndication> tags) Indication Amount - Pricing <Amount> (within<PricingIndication> tags) Indication Increment - Pricing <Increment>(within <PricingIndication> tags) Indication Unit - Pricing <Unit>(within <PricingIndication> tags) Indication Type of service <Service>requested SubscriberInfo <SourceAlternate type=″subscriber″> CustomerId<CustomerId> DeviceId <DeviceId> Data Rate <DataRate> Number of Channels<NumberOfChannels> Bandwidth <Bandwidth> Codec <Codec> Quality ofService <QualityOfService> Quality of Service <QoSClass> Class AnswerSeizure <AnswerSeizureRatio> Ratio (ASR) (within <QualityOfService>tags) Mean Hold <MeanHoldTime> Time (MHT) (within <QualityOfService>tags) Post Dial <PostDialDelay> Delay (PDD) (within <QualityOfService>tags) OSP Version <OSPVersion> Call Identifier <CallId> Delay <Delay>Jitter <Jitter> Packet Loss <PackLoss>Peering Authorization Response 315

The Peering Authorization Response 315 from the Peering Authority(Clearinghouse 300) to the Source IP Network 100 may include thefollowing information listed in Table 3 below:

TABLE 3 Peering Authorization Response Information Information ElementNote Date and time Date/Time of authorization response. of response CallIdentifier Any unique identifier for the call or session. Group Id Sameas ConferenceID in H.323. Calls with unique CallIds can share a commonGroupID. i.e a conference call. Calling party Unique identifier for thecalling party, i.e.: E.164 number, sip uri, IP address, name resolved byDomain Name Server (DNS) to an IP address. Source network IP address orname which can be resolved using DNS Source device IP address or namewhich can be resolved using DNS. Source trunk String value with trunkgroup. This value may or may group not include circuit ID. Receivingparty Unique identifier for the receiving party, i.e.: E.164 number, sipuri, IP address, name resolved by DNS to an IP address. ApplicationApplication requested by the source network and served by thedestination network. This includes any application provided as a servicesuch as a gaming or video streaming from a specific web camera. FileFile requested by the source network and served by the destinationnetwork. This includes data files, ring tones, audio files or videofiles. Destination IP address or name which can be resolved using DNS.signaling address Destination trunk String value with trunk group. Thisvalue may or may group not include circuit ID. Destination h323-Q931,h323-LRQ, SIP, IAX Protocol OSP version Most recent version of OSPprotocol supported by the destination. 0.0.0, 1.4.3, 2.1.1, 4.1.1Authorization Described in the next section. TokenPeering Authorization Token

For non-repudiation of peering authorization and settlement services,the peering authorization token usually must be digitally signed withthe private key of the Peering Authority (Clearinghouse 300). Thepeering authorization token may be encoded, encrypted or plain text. Thetoken defines what type of service, quantity of service, quality ofservice and pricing has been authorized by the Peering Authority.

The peering authorization token returned from a Peering Authority(Clearinghouse 300) may include the information elements in Table 4below.

TABLE 4 Peering Authorization Token Information Information ElementDescription Version Version of the token. Random number Any randomlygenerated number. Transaction ID A unique number identified for the callor peering generated by the session. settlement server Contact IDIdentifies the Peering Authority. Valid after time Call or session mustbe established after this time. Valid until time Call or session must beestablished before this time. Calling Party Unique identifier for thecalling party, i.e.: E.164 number, sip uri, IP address, name resolved byDomain Name Server (DNS) to an IP address. Calling Party A set ofinformation (Calling Party Name, Calling Identification Party Authorityand Calling Party Verification) which identifies the calling party.Calling Party Name or text description of the calling party. For Namecalling parties from the PSTN, this value would typically be name from aLine Interface Database (LIDB) or Calling Name (CNAM) database thatcorresponds to the calling party's telephone number. Calling PartyOrganization of the calling party. Organization Calling Party First nameof calling party. First Name Calling Party Last name of calling party.Last Name Calling Party Street address of calling party. Street NameCalling Party Street number of calling party. Street Number CallingParty Additional address information, such as apartment Address2 orsuite of calling party. Calling Party Postal code of calling party.Postal Code Calling Party City of calling party. City Calling PartyState of calling party. State Calling Party Country of calling party.Country Calling Party Unique number identifying the calling party. IdNumber Calling Party Website of calling party. Website Calling PartyUniform Resource Identifier of calling party. URI Calling Party Name ortext description of the authority that Authority authenticated andverified the calling party identification for the peering authorizationrequest. Calling Party Integer value which indicates the level ofVerification verification of the calling party identification. 00 - Noverification 01 - Based on the calling party's telephone number 02 -Based on source device IP address 03 - Combination of 01 and 02 04 -Based on password of calling party 05 - Combination of 01 and 04 07 -Combination of 01, 02 and 04 08 - Based on the SSL/TLS clientauthentication of the calling party. Source Network A set of information(Source Network Name, Identification Source Network Authority and SourceNetwork Verification) which identifies the operator of the sourcenetwork. Source Network Name or text description of the source networkName operator. Source Network Organization of the Source Network.Organization Source Network Street address of Source Network. StreetName Source Network Street number of Source Network. Street NumberSource Network Additional address information, such as apartmentAddress2 or suite of Source Network. Source Network Postal code ofSource Network. Postal Code Source Network City of Source Network. CitySource Network State of Source Network. State Source Network Country ofSource Network. Country Source Network Unique number identifying theSource Network. Id Number Source Network Website of Source Network.Website Source Network Uniform Resource Identifier of Source Network.URI Source Network Name or text description of the authority thatAuthority authenticated and verified the source network identificationfor the peering authorization request. Source Network Integer valuewhich indicates the level of Verification verification of the sourcenetwork identification. 00 - No verification 01 - Based on the callingparty's telephone number 02 - Based on source device IP address 03 -Combination of 01 and 02 04 - Based on password of calling party 05 -Combination of 01 and 04 07 - Combination of 01, 02 and 04 08 - Based onthe SSL/TLS client authentication of the calling party. If theVerification technique is unknown, then the Verification value should beempty. Receiving party Unique identifier for the receiving party, i.e.:E.164 number, sip uri, IP address, name resolved by Domain Name Server(DNS) to an IP address. Application Application requested by the sourcenetwork and served by the destination network. This includes anyapplication provided as a service such as a gaming or video streamingfrom a specific web camera. File File requested by the source networkand served by the destination network. This includes data files, ringtones, audio files or video files. Amount - Usage Amount of usageauthorized Detail Increment - Usage Number of units per increment DetailUnits - Usage Seconds, packets, bytes, pages, calls Detail Currency -Pricing Currency of transaction, i.e. USD Indication Setup - PricingFixed price per call Indication Amount - Pricing Price per incrementIndication Increment - Pricing Units per increment Indication Units -Pricing Seconds, packets, bytes, pages, calls Indication Type of serviceService requested - voice, video, data. Default is voice. Destinationaddress Used for look ahead routing Destination trunk Used for lookahead routing group Destination protocol Used for look ahead routing OSPVersion Used for look ahead routing Call Identifier Unique identifierfor the call Group ID (i.e for Same as ConferenceID in H.323. Calls withunique conference call CallIds can share a common GroupID. i.e aadmission) conference call. Data Rate Data rate requested for the callor peering session Number of Channels Number of channels requestedBandwidth Amount of bandwidth reserved Codec Compression/decompressionalgorithm Quality of Service Level of service quality requested Qualityof Service Class of service quality requested Class Answer Seizure <ASR>(within <QualityOfService> tags) Ratio (ASR) Mean Hold <MHT> (within<QualityOfService> tags) Time (MHT) Post Dial <PDD> (within<QualityOfService> tags) Delay (PDD)Peering Authorization Token—XML Mapping

Table 5 below maps peering authorization token information elements toeXtensible Markup Language (XML) tags and ASCII tags.

TABLE 5 Peering Authorization Token - XML Tags Information ASCII ElementXML Tag Tag Version <Version> V Random Number <TokenInfo random=′30113′>r Transaction ID <TransactionId> I ContactID <ContactId> x Valid AfterTime <ValidAfter> a Valid Until Time <ValidUntil> u Calling Number<SourceInfo> c Calling Party <Name> within <CallingPartyId> tags J NameCalling Party <Organization> within <CallingPartyId> ! Organization tagsCalling Party <FirstName> within <CallingPartyId> $ First Name tagsCalling Party <LastName> within <CallingPartyId> tags } Last NameCalling Party <StreetName> within <CallingPartyId> ′ Street Name tagsCalling Party <StreetNumber> within <CallingPartyId> / Street Numbertags Calling Party <Address2> within <CallingPartyId> tags \ Address2Calling Party <PostalCode> within <CallingPartyId> ) Postal Code tagsCalling Party <City> within <CallingPartyId> tags % City Calling Party<State> within <CallingPartyId> tags ? State Calling Party <Country>within <CallingPartyId> tags & Country Calling Party <IdNumber> within<CallingPartyId> tags ] Id Number Calling Party <Authority> within<CallingPartyId> tags j Authority Calling Party <Verification> within<CallingPartyId> k Verification tags Source Network <Name> within<SourceNetworkId> tags S Name Source Network <Organization> within <Organization <SourceNetworkId> tags Source Network <StreetName> within ;Street Name <SourceNetworkId> tags Source Network <StreetNumber> within— Street Number <SourceNetworkId> tags Source Network <Address2> within<SourceNetworkId> = Address2 tags Source Network <PostalCode> within<SourceNetworkId> . Postal Code tags Source Network <City> within<SourceNetworkId> tags ( City Source Network <State> within<SourceNetworkId> tags > State Source Network <Country> within<SourceNetworkId> {circumflex over ( )} Country tags Source Network<IdNumber> within <SourceNetworkId> [ Id Number tags Source Network<Authority> within <SourceNetworkId> A Authority tags Source Network<Verification> P Verification Called Number <DestinationInfo> CApplication <Application> g File <File> F Amount - Usage <Amount>(within <UsageDetail> tags) M Detail Increment - Usage <Increment>(within <UsageDetail> tags) n Detail Units - Usage <Units> (within<UsageDetail> tags) U Detail Currency - Pricing <Currency> (within<PricingIndication> y Indication tags) Setup - Pricing <Setup> (within<PricingIndication> tags) p Indication Amount - Pricing <Amount> (within<PricingIndication> m Indication tags) Increment - Pricing <Increment>(within <PricingIndication> N Indication tags) Units - Pricing <Units>(within <PricingIndication> tags) f Indication Type of service <Service>s Destination address <DestinationAlternate type=’transport’> d for lookahead routing Destination trunk <DestinationAlternate type=’network’> egroup for look ahead routing Destination protocol <DestinationProtocol>D for look ahead routing OSP Version <OSPVersion> o for look aheadrouting Call Identifier <CallId encoding=′base64′> i CallId may or maynot be encoded Look Ahead Tag <LookAhead> K Group ID <GroupId> G DataRate <DataRate> R Number of Channels <NumberOfChannels> b Bandwidth<Bandwidth> B Codec <Codec> E Quality of Service <QualityOfService> QQuality of Service <QoSClass> q ClassAccounting

After a VoIP call, or a peering session, is completed it must beaccounted. The call or peering session details must be reported to thesettlement clearing or peering authority (Clearinghouse 300). Table 5below lists Information Elements which should be included in a call orsession detail record. The XML format is provided for informationelements which have not been defined previously in this document.

TABLE 5 Call or Session Record Information Information Element Note TimeStamp Time/Date of the Message Role Source, Destination, OtherTransaction ID Same as Peering Authorization Response Call ID Same asPeering Authorization Response Source Network Same as PeeringAuthorization Response Source Device Same as Peering AuthorizationResponse Source Trunk Same as Peering Authorization Response GroupCalling Number Same as Peering Authorization Response Subscriber Same asPeering Authorization Response information Called Number Same as PeeringAuthorization Response Application Same as Peering AuthorizationResponse File Same as Peering Authorization Response Destination NetworkSame as Peering Authorization Response Destination Trunk Same as PeeringAuthorization Response Group Amount - Usage Amount of usage (callduration) Indication Increment - Usage Number of units per incrementIndication Units - Usage Seconds, packets, bytes, pages, callsIndication Currency - Pricing Currency of transaction, i.e. USDIndication Amount - Pricing Price per increment Indication Increment -Pricing Units per increment Indication Units - Pricing Seconds, packets,bytes, pages, calls Indication Call Start Time The time when the callingparty starts the call, i.e. after last digit is dialed or when the sendbutton is pushed. <StartTime> Call Alerting Time When the calledtelephone begins ringing <AlertTime> Call Connect Time When the callingand receiving party connect <ConnectTime> Call End Time When the callends <EndTime> Post Dial Delay Time elapsed between Call Alert time andCall Start Time. <PostDialDelay> within <UsageDetail> tags Statistics Asdefined in ETSI TS 101 321 V4.1.1 Customer Id Same as PeeringAuthorization Response Device ID Same as Peering Authorization ResponseGroupID Same as Peering Authorization Response Data Rate Data raterequested for VoIP call or IP session. Number of Channels Number ofchannels requested for IP session. Bandwidth Amount of bandwidthreserved for IP session. Codec Compression/decompression algorithmrequested. Quality of Service Level of service quality requested.Quality of Service Class of service quality requested. Class TerminationReason why the call was disconnected. <TCCode> Cause Code Source ofRelease IP address of device which terminated the VoIP call Complete orIP session <ReleaseSource>

Referring now to FIG. 4a , this Figure illustrates an inter-IP networkpeering scenario which illustrates peering authorization request andresponse messages and the peering authorization token. The example callscenario in FIG. 4a illustrates a call from an end user on the NationalBank Corp Network 500 to a receiving party 610 with telephone number2564286000 served by an external network 600. This call scenariorequires interconnection, or peering, between the National Bank CorpNetwork 500 and a destination network 600.

The call begins with the calling party 510, John Doe in the AccountsPayable department of National Bank Corp. The calling party has atraditional circuit switched telephone with telephone number 4045266060.The telephone is connected via Circuit 1 of Trunk Group S 520 to a SIPgateway 530 having an IP address 1.1.1.1. SIP, or Session InitiationProtocol, is an IP protocol used transmitting voice, video and othercommunications over IP networks as is known to one of ordinary skill inthe art. The SIP gateway 530 is registered with the SIP Proxy 540 withIP address 1.1.1.2. The SIP Proxy 540 performs the roll of the CallControl Point 110 that is illustrated in FIG. 2 a.

When the calling party 510 calls the telephone number 2564286000, a callsetup message is sent from SIP Gateway 530 to the SIP Proxy 540. The SIPProxy determines to call either Destination Network 1 600 or DestinationNetwork 2 605. Table 7 below summarizes the routing data and peeringcriteria for completing the call from the SIP Proxy 540 to a destinationnetwork which can complete the call to the receiving party 610.

TABLE 7 Summary of Routing Data and Peering Criteria for CompletingPeering Criteria Possible Destinations Network 1 Network 2 IP Address2.2.2.2 3.3.3.3 Trunk Group and Circuit ID D1: A D2: B Peering Rate 0.1010 Billing Increment 60 60 Billing Units seconds seconds Currency USDJPY Bandwidth required in kb/second 64 64 Minimum acceptable AnswerSeizure Ratio 0.5 0.5 Minimum acceptable Mean Hold Time 120 120 Minimumacceptable Post Dial Delay 2 2

Table 7 above includes routing information, such as the IP addresses ofSIP Gateways 630 and 635 and trunk group—circuit details 620 and 625connecting to the receiving party telephone 610. Table 7 also includespeering rate information which can be used to bill for the peeringsession. The peering rate for 64 kbs of bandwidth to terminate a voicecall on Network 1—600 is $0.10 USD per 60 seconds. For Network 2—605 thepeering rate for the same service is 10 JPY per 60 seconds. Table 7 alsoincludes minimum acceptable quality of service in terms of AnswerSeizure Ratio, Mean Hold Time and Post Dial Delay.

Before sending a call setup to either Destination Network 1—600 orDestination Network 2—605, the SIP Proxy 540 sends a peeringauthorization request 710 to the Peering Authority, ACME SettlementClearinghouse 700. Below is the peering authorization request message710. It is an eXtensible Markup Language (XML) message transmitted viaHyper Text Transport Protocol (HTTP).

POST /osp HTTP/1.1 Host: 172.16.4.78 content-type: text/plainContent-Length: 745 Connection: Keep-Alive <?xml version=“1.0”?><Message messageId=“127821” random=“12782”> <AuthorizationRequestcomponentId=“127820”> <Timestamp>2004-12-06T21:58:03Z</Timestamp><CallId encoding=“base64”>vIAeOEcIEdmUff2oaL5HDA==</CallId> <SourceInfotype=“e164”>4045266060</SourceInfo> <CallingPartyId> <Name>John Doe -Accounts Payable<Name> <Authority>National Bank Corp<Authority><Verification>02<Verification> </CallingPartyId> <DeviceInfotype=“transport”>[1.1.1.1]</DeviceInfo> <SourceAlternatetype=“network”>TrunkGrpS:1</SourceAlternate> <SourceAlternatetype=“transport”>[1.1.1.2]</SourceAlternate> <DestinationInfotype=“e164”>2564286000</DestinationInfo> <Destination><DestinationAlternate type=“network”>TrunkGrpD1:A</DestinationAlternate><DestinationAlternate type=“transport”>[2.2.2.2]</DestinationAlternate><PricingIndication> <Currency>USD</Currency> <Amount>0.10</Amount><Increment>60</Increment> <Unit>s</Unit> </PricingIndication><Bandwidth>64</Bandwidth> <QualityOfService><AnswerSeizureRatio>0.5</AnswerSeizureRatio><MeanHoldTime>120</MeanHoldTime> <PostDialDelay>2</PostDialDelay></QualityOfService> </Destination> <Destination> <DestinationAlternatetype=“network”>TrunkGrpD2:B</DestinationAlternate> <DestinationAlternatetype=“transport”>[3.3.3.3]</DestinationAlternate> <PricingIndication><Currency>JPY</Currency> <Amount>10</Amount> <Increment>60</Increment><Unit>s</Unit> </PricingIndication> <Bandwidth>64</Bandwidth><QualityOfService> <AnswerSeizureRatio>0.5</AnswerSeizureRatio><MeanHoldTime>120</MeanHoldTime> <PostDialDelay>2</PostDialDelay></QualityOfService> </Destination> <CustomerIdcritical=“False”>1000</CustomerId> <DeviceIdcritical=“False”>1000</DeviceId> <Service/> </AuthorizationRequest></Message>

In the peering authorization request 710 above, separate pricing,bandwidth and quality of service criteria is specified for eachdestination within each set of <Destination> tags. A global set ofpeering critiera may be applied to all destinations by including thecriteria information within the request message, but outside of the<destination> tags.

When the ACME Settlement Clearinghouse 700 receives the peeringauthorization request 710, it may perform the following functions:

-   -   1. Authenticate the source network 500. Authentication of the        source network 500 may be accomplished using various techniques        such as authenticating the IP address of the source Call Control        Point 540 or using Secure Sockets Layer/Transport Layer Security        (SSL/TLS) client authentication.    -   2. Determine if the operator of the source network 500 may        originate a call to the receiving party 610. For example, the        source network 500 may be denied authorization because the        operator of the source network has insufficient credit with the        Settlement Clearinghouse 700.    -   3. Determine if the calling party 510 or source device may        originate a call to the receiving party 610. For example, the        source device may be blocked from originating calls due to        specific Settlement Clearinghouse 300 policies.    -   4. Determine if the destination networks 600 and 605 may        terminate calls. For example, authorization of calls may be        denied to destination network by the Settlement Clearinghouse        700 based on business policies.    -   5. Determine if the destination devices or Call Control Point        630 or 635 may terminate the call. For example, authorization of        calls to a destination device may be denied if historical        quality of service statistics (ASR, MHT, PDD, Delay, Jitter or        PackLoss) for VoIP calls using the device are less than the        minimum quality of service levels defined in the peering        authorization request 710.    -   6. Determine if the Settlement Clearinghouse 700 can use the        pricing information provided in the peering authorization        request:        -   a. Is Currency acceptable?        -   b. Is Setup pricing acceptable?        -   c. Is Amount acceptable?        -   d. Is Billing Increment acceptable?        -   e. Is Unit acceptable?    -   7. If the call is authorized to the one or more of the        destination networks 600, 605, create a peering authorization        token with a specified usage amount based on business policies        or criteria such as:        -   a. Credit balance of the Source Network 500 operator.        -   b. Price of interconnect.    -   8. Create call detail record, at the ACME Settlement        Clearinghouse 700, which documents the details of the peering        authorization request including interconnect pricing        information.    -   9. Send a peering authorization response message to the SIP        Proxy 540 of the source network 500.

Below is the peering authorization response message 720 illustrated inFIG. 4b . This message sent from the ACME Settlement Clearinghouse 700to the SIP Proxy 540 and authorizes the peering request to bothDestination Network 1—600 and Destination Network 2—605 for 14400seconds:

<?xml version=‘1.0’?> <Message messageId=‘127821’ random=‘12782’><AuthorizationResponse componentId=‘127820’><Timestamp>2004-12-06T21:58:03Z</Timestamp> <Status><Description>SUCCESS</Description> <Code>200</Code> </Status><TransactionId>4733131489186097309</TransactionId> <Destination> <CallIdencoding=‘base64’>vIAeOEcIEdmUff2oaL5HDA==</CallId> <DestinationInfotype=‘e164’>2564286000</DestinationInfo><DestinationSignalAddress>[2.2.2.2]</DestinationSignalAddress> <Tokenencoding=‘base64’>lcnZlcjBcMA0GCSqGSIb3DQEBAQUAA0sAMEgCQQDxnhsFeNRCIV964IxAxS0V0IQxK+dDlED6vn+eVcEHdE0DbbNOgT9vflIjXUVt3NyER3WXCXQtQoOvDBPXIHqtIfAgMBAAEwDQYJKoZIhvcNAQEEBQADQQCbizVxrw9HaIYB5MawCrqpTS4xV7p4hOu+m7rv6qUJHU6eHFw911P1iubOAyMC+s46hE7c1C8IRgYRxEclzfudMYGjMIGgAgEBMDwwNzEhMB8GA1UEAxMYU3VuRTQ1MC05LnRyYW5zbmV4dXMuY29tMRIwEAYDVQQKEwlPU1BTZXJ2ZXICAQEwDAYIKoZIhvcNAgUFADANBgkqhkiG9w0BAQEFAARAfNbeo5Y19qFSeITbJQO8Lq8y7DwKukhXY4FTEJFc0zUPq+az9rkaGrjzW6+0+3lQlAftXekTteP6RilVbfX6nw== </Token> <UsageDetail><Amount>14400</Amount> <Increment>1</Increment> <Service/><Unit>s</Unit> </UsageDetail><ValidAfter>2004-12-06T21:53:03Z</ValidAfter><ValidUntil>2004-12-06T22:03:03Z</ValidUntil> <DestinationProtocolcritical=‘False’>IAX</DestinationProtocol> <OSPVersioncritical=‘False’>1.4.3</OSPVersion> <SourceInfo type=‘e164’critical=‘False’>4045266060</SourceInfo> <DestinationAlternatetype=‘network’critical=‘False’>TrunkGrpD1:A</DestinationAlternate></Destination> <Destination> <CallIdencoding=‘base64’>vIAeOEcIEdmUff2oaL5HDA==</CallId> <DestinationInfotype=‘e164’>2564286000</DestinationInfo><DestinationSignalAddress>[3.3.3.3]</DestinationSignalAddress> <Tokenencoding=‘base64’>lcnZlcjBcMA0GCSqGSIb3DQEBAQUAA0sAMEgCQQDxnhsFeNRCIV964IxAxS0V0IQxK+dDlED6vn+eVcEHdE0DbbNOgT9vflIjXUVt3NyER3WXCXQtQoOvDBPXIHqtIfAgMBAAEwDQYJKoZIhvcNAQEEBQADQQCbizVxrw9HaIYB5MawCrqpTS4xV7p4hOu+m7rv6qUJHU6eHFw911P1iubOAyMC+s46hE7c1C8IRgYRxEclzfudMYGjMIGgAgEBMDwwNzEhMB8GA1UEAxMYU3VuRTQ1MC05LnRyYW5zbmV4dXMuY29tMRIwEAYDVQQKEwlPU1BTZXJ2ZXICAQEwDAYIKoZIhvcNAgUFADANBgkqhkiG9w0BAQEFAARAfNbeo5Y19qFSeITbJQO8Lq8y7DwKukhXY4FTEJFc0zUPq+az9rkaGrjzW6+0+3lQlAftXekTteP6RilVbfX6nw== </Token> <UsageDetail><Amount>14400</Amount> <Increment>1</Increment> <Service/><Unit>s</Unit> </UsageDetail><ValidAfter>2004-12-06T21:53:03Z</ValidAfter><ValidUntil>2004-12-06T22:03:03Z</ValidUntil> <DestinationProtocolcritical=‘False’>SIP</DestinationProtocol> <OSPVersioncritical=‘False’>2.1.1</OSPVersion> <SourceInfo type=‘e164’critical=‘False’>4045266060</SourceInfo> <DestinationAlternatetype=‘network’critical=‘False’>TrunkGrpD2:B</DestinationAlternate></Destination> </AuthorizationResponse> </Message>

At least one important information element in the peering authorizationresponse message 720 is the peering authorization token. The sourcenetwork Call Control Point 110 (see FIG. 2a ), or the SIP Proxy 540 inFIG. 4b will include this token in its call setup to the destinationnetwork 600, 605 for access permission to complete the call. The tokenis base64 encoded in these examples and is not readable. However, belowis a decoded version of the token for Destination Network 1—600. Thetoken lists all of the peering criteria defined in the peeringauthorization request. Also, the token includes information about thesource network identification. This information, combined with thecalling party information, provides the receiving party with informationabout the calling party and the organization from which the calloriginated.

<?xml version=‘1.0’?> <TokenInfo random=‘30113’> <SourceInfotype=‘e164’>4045266060</SourceInfo> <CallingParty> <Identification>JohnDoe - Accounts Payable<Identification> <Authority>National BankCorp<Authority> <Verification>02<Verification> </CallingParty><SourceNetwork> <Identification>National Bank Corp<Identification><Authority>ACME Settlement Clearinghouse<Authority><Verification>07<Verification> </SourceNetwork> <DestinationInfotype=‘e164’>2564286000</DestinationInfo> <CallIdencoding=‘base64’>NjMzMDYxMzgzMDMxMzA2MTMyMzAzMDAwMDA=</CallId><ValidAfter>2004-12-02T20:37:58Z</ValidAfter><ValidUntil>2004-12-02T20:47:58Z</ValidUntil><TransactionId>4733140074823188489</TransactionId> <UsageDetail><Amount>14400</Amount> <Increment>1</Increment> <Service/><Unit>s</Unit> </UsageDetail> <PricingIndication><Currency>USD</Currency> <Setup>0<Setup> <Amount>0.10</Amount><Increment>60</Increment> <Unit>s</Unit> </PricingIndication><Bandwidth>64</Bandwidth> </TokenInfo>Example of Peering Authorization Token in Text Format

Peering authorization tokens may be in any format, not just an XMLformat. Below is the peering authorization reformatted in AmericanStandard Code for Information Interchange (ASCII) text format:

V=1 r=30113 c=4045266060 J=John Doe - Accounts Payable j=National BankCorp k=02 S=National Bank Corp A=ACME Settlement Clearinghouse P=07C=2564286000 i=NjMzMDYxMzgzMDMxMzA2MTMyMzAzMDAwMDA=a=2004-12-02T20:52:30Z u=2004-12-02T21:02:30Z I=4733144047667937289M=14400 n=1 s= U=s p=USD e=0 p=0.1 N=60 f=s B=64

When the source SIP Proxy 540 receives the peering authorization request720 from the Clearinghouse 700, it extracts the peering authorizationtoken and includes the token in the call setup message 730 to thedestination network as illustrated in FIG. 4c . Specifically, the sourceSIP Proxy 540 sends a call setup message 730 to the SIP Gateway 630 inDestination Network 1 600. The token for Destination Network 1—600 isincluded in the call setup message 730.

The SIP Gateway 630 of the Destination Network 1—600 performs the stepsbelow to validate that the call has been authorized by the ACMESettlement Clearinghouse and that the interconnect, or peering, termsare acceptable:

-   -   1. Validate the digital signature of the token using the public        key of the ACME Settlement Clearinghouse certificate authority.    -   2. Validate the service authorized in the token is acceptable.        If no service is defined in the token, it assumed to be voice.    -   3. Validate the quality of service level requested is        acceptable. If not, the call must be rejected.    -   4. Validate the interconnect or peering price information.        -   a. Amount        -   b. Setup        -   c. Increment        -   d. Units        -   e. Currency        -   If the price information is not acceptable, the call should            not be accepted.    -   5. Are the Calling Party Id and Source Network Id information        acceptable? For example, the destination network may want to        block the call based on the Authority or Verification level.        If the token is valid and its terms (interconnect rate, required        quality of service, etc.) are acceptable, the destination SIP        Gateway 630 accepts the call setup message 730 and completes the        call to receiving party over Trunk Group D1 Circuit A 620 to the        receiving party 610. If the call setup 730 was rejected by the        destination SIP Gateway 630, the source SIP Proxy 540 would make        a second attempt to complete the call destination SIP Gateway        635 of Destination Network 2—605. SIP Gateway 635 would also        validate the token and assess if the peering terms were        acceptable before accepting the call.        Accounting

At the end of an interconnect VoIP call, or peering session, both thesource and destination networks 600, 605 should report accountingrecords to the peering authority 700. As illustrated in FIG. 4d , thesource SIP Proxy 540 sends a call record 740 to the ACME SettlementClearinghouse 700 and the destination SIP Gateway 630 also sends a callrecord 740 to the ACME Settlement Clearinghouse 700. Below is an exampleof peering accounting message 740 from the Source SIP Proxy 540 to theACME Settlement Clearinghouse 700:

POST /osp HTTP/1.1 Host: 172.16.4.78 content-type: text/plainContent-Length: 1837 Connection: Keep-Alive <?xml version=“1.0”?><Message messageId=“47331314891860973093” random=“2730”><UsageIndication componentId=“47331314891860973092”><Timestamp>2004-12-06T21:58:53Z</Timestamp> <Role>source</Role><TransactionId>4733131489186097309</TransactionId> <CallIdencoding=“base64”>vIAeOEcIEdmUff2oaL5HDA==</CallId> <SourceInfotype=“e164”>4045266060</SourceInfo> <DeviceInfotype=“transport”>[1.1.1.1]</DeviceInfo> <SourceAlternate type=“network”>TrunkGrpS:1</SourceAlternate> <SourceAlternatetype=“transport”>[1.1.1.2]</SourceAlternate> <DestinationInfotype=“e164”>2564286000</DestinationInfo> <DestinationAlternatetype=“transport”>[2.2.2.2]</DestinationAlternate> <DestinationAlternatetype=“network”>TrunkGrpD1:A</DestinationAlternate> <Group><GroupId>BC801E38-4708-11D9-947D-FDA868BE470C</GroupId> </Group><UsageDetail> <Service/> <Amount>300</Amount> <Increment>1</Increment><Unit>s</Unit> <StartTime>2004-12-06T21:58:03Z</StartTime><EndTime>2004-12-06T21:58:25Z</EndTime><ConnectTime>2004-12-06T21:58:05Z</ConnectTime><ReleaseSource>0</ReleaseSource> </UsageDetail> <PricingIndication><Currency>USD</Currency> <Setup>0<Setup> <Amount>0.10</Amount><Increment>60</Increment> <Unit>s</Unit> </PricingIndication><CustomerId critical=“False”>1000</CustomerId> <DeviceIdcritical=“Fa1se”>1000</DeviceId> <FailureReason>1016</FailureReason><Statistics critical=“False> <LossSent critical=“False”> <Packetscritical=“False”>0</Packets> <Fraction critical=“False”>0</Fraction></LossSent> <LossReceived critical=“False”> <Packetscritical=“False”>0</Packets> <Fraction critical=“False”>0</Fraction></LossReceived> </Statistics> </UsageIndication> </Message>Settlement

When the call, or peering session, accounting records 740 are collected,the ACME Settlement Clearinghouse 700 may then perform additionalservices for the source and destination peers such as interconnectbilling, determination of net settlement 40, payments among the peering,execution of any settlement payment transaction, analysis and reportingof inter-peer traffic statistics.

Other Applications of Peering in IP Communications

One of ordinary skill in the art of IP communications recognizes thatthe peering technique described above for VoIP applications could alsobe used in other IP applications that require peering between twonetworks, access control and accounting. A video call, which is astraight-forward extension of the VoIP call scenario described indetail, is an application which could benefit from the techniquesdescribed. Other peering applications include data file downloads,interactive gaming, application services or content brokering.

The following example illustrates how the invention can be used forapplications other than VoIP. Assume the source network is a serviceprovider offering on-line movie services to its end user subscribers. Ifthe source network does not have the movie content requested by asubscriber (calling party), the source network can send a peeringrequest to a content broker (clearinghouse) to request access to thenetwork distributor of the requested movie. In this example, the networkdistributor is analogous to the destination network in the VoIP callscenario. The requested movie, or application, is analogous to thereceiving party. The content broker would approve the peering requestand create a peering token specifying the movie and requested bandwidthfor the movie media stream.

The source network would then forward the peering token to the networkdistributor. The network distributor would then validate the peeringtoken, identify the movie (application) requested and then provide themovie media stream to the source network. The source network would thenredirect the movie stream to its end user. At the end of the movie mediastream, the source network and network distributor would send theiraccounting records to the content broker who would facilitate billingand payment from the source network to the network distributor.

It should be understood that the foregoing relates only to illustratethe embodiments of the invention, and that numerous changes may be madetherein without departing from the scope and spirit of the invention asdefined by the following claims.

What is claimed is:
 1. A method for securely authorizing a voice overinternet protocol (VOIP) call between peers of computer networks:identifying peering criteria in a voice over internet protocol (VOIP)call authorization request; sending the VOIP call authorization requestcomprising the peering criteria to a peering authority; evaluating thepeering criteria with the peering authority in which the peeringauthority compares prices in one or more pricing tables with the pricefor each individual VOIP call set forth in the peering criteria; andgenerating one or more tokens corresponding to one or more computernetworks for completing the VOIP call based on the peering criteriausing the one or more tokens, the method further comprises identifyingthe one or more second peer computer networks that are capable ofcompleting the VOIP call originating from a first peer computer network.2. The method of claim 1, wherein the peering criteria comprises priceinformation for the VOIP call.
 3. The method of claim 1, furthercomprising determining if a VOIP call originating from the first peercomputer network must be completed by one or more second peer computernetworks that are separate from the first computer network.
 4. Themethod of claim 1 further comprising identifying peering criteriacomprising at least one of bandwidth and network quality of service inVOIP call authorization request.
 5. The method of claim 1, furthercomprising completing the VOIP call with a token.
 6. The method of claim5, further comprising sending one or more tokens to a first peercomputer network; and sending a call setup request comprising a tokenfrom the first peer computer network to a second computer network. 7.The method of claim 1, wherein identifying one or more second peercomputer networks further comprises reviewing available second peercomputer networks stored in a routing table.
 8. The method of claim 1,wherein identifying one or more second peer computer networks furthercomprises discovering second peer computer networks using routediscovery protocols.
 9. The method of claim 1, wherein evaluating thepeering criteria further comprises identifying a type of servicerequested in the peering criteria.
 10. The method of claim 1, whereinevaluating the peering criteria further comprises determining if pricingassociated with the VOIP are acceptable by one or more second peercomputer networks.
 11. The method of claim 1, wherein evaluating thepeering criteria further comprises determining if quality of serviceassociated with the peering criteria are acceptable by one or moresecond peer computer networks.
 12. A system for securely authorizing avoice over internet telephony call between devices of internet telephonynetworks comprising: a first call point control device of a firstcomputer network; the first call point control device determiningpeering criteria for the voice over internet protocol (VOIP) call; and apeering authority coupled to the call point control device, forreceiving the peering criteria and evaluating the peering criteria, forgenerating one or more tokens corresponding to the one or more secondpeer computer networks based on the evaluation of the peering criteria,each token establishing price for the VOIP call in accordance with thepeering criteria, the first call point control device selecting a tokenand contacting a second call point control device on a second computernetwork associated with the selected token for completing the VOIP callwith the token.
 13. The system of claim 12, wherein the first call pointcontrol device identifies one or more second peer computer networks thatare capable of completing a VOIP call from a first telephone to a secondtelephone.
 14. The system of claim 12, wherein the peering criteriacomprises a price.
 15. The system of claim 12, wherein the first callpoint control device determines if VOIP call originating from a firsttelephone to a second telephone must be completed by one or more secondcomputer networks that are separate from the first computer network. 16.The system of claim 12, wherein the peering criteria comprises at leastone of bandwidth and a network quality of service for the VOIP call. 17.The system of claim 12, wherein each token further establishes at leastone of bandwidth and quality of service for the VOIP call in accordancewith the peering criteria.
 18. The system of claim 12, wherein thepeering authority and first and second call point control devicescomprise computer servers with internet protocol (IP) addresses.
 19. Thesystem of claim 12, wherein the second call point control devicereceives the selected token and determines if the token is valid.